path: root/fdk-aac/libSYS/include/FDK_audio.h

/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android

© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.

 1.    INTRODUCTION
The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
scheme for digital audio. This FDK AAC Codec software is intended to be used on
a wide variety of Android devices.

AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
general perceptual audio codecs. AAC-ELD is considered the best-performing
full-bandwidth communications codec by independent studies and is widely
deployed. AAC has been standardized by ISO and IEC as part of the MPEG
specifications.

Patent licenses for necessary patent claims for the FDK AAC Codec (including
those of Fraunhofer) may be obtained through Via Licensing
(www.vialicensing.com) or through the respective patent owners individually for
the purpose of encoding or decoding bit streams in products that are compliant
with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
Android devices already license these patent claims through Via Licensing or
directly from the patent owners, and therefore FDK AAC Codec software may
already be covered under those patent licenses when it is used for those
licensed purposes only.

Commercially-licensed AAC software libraries, including floating-point versions
with enhanced sound quality, are also available from Fraunhofer. Users are
encouraged to check the Fraunhofer website for additional applications
information and documentation.

2.    COPYRIGHT LICENSE

Redistribution and use in source and binary forms, with or without modification,
are permitted without payment of copyright license fees provided that you
satisfy the following conditions:

You must retain the complete text of this software license in redistributions of
the FDK AAC Codec or your modifications thereto in source code form.

You must retain the complete text of this software license in the documentation
and/or other materials provided with redistributions of the FDK AAC Codec or
your modifications thereto in binary form. You must make available free of
charge copies of the complete source code of the FDK AAC Codec and your
modifications thereto to recipients of copies in binary form.

The name of Fraunhofer may not be used to endorse or promote products derived
from this library without prior written permission.

You may not charge copyright license fees for anyone to use, copy or distribute
the FDK AAC Codec software or your modifications thereto.

Your modified versions of the FDK AAC Codec must carry prominent notices stating
that you changed the software and the date of any change. For modified versions
of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
AAC Codec Library for Android."

3.    NO PATENT LICENSE

NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
Fraunhofer provides no warranty of patent non-infringement with respect to this
software.

You may use this FDK AAC Codec software or modifications thereto only for
purposes that are authorized by appropriate patent licenses.

4.    DISCLAIMER

This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
including but not limited to the implied warranties of merchantability and
fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
or consequential damages, including but not limited to procurement of substitute
goods or services; loss of use, data, or profits, or business interruption,
however caused and on any theory of liability, whether in contract, strict
liability, or tort (including negligence), arising in any way out of the use of
this software, even if advised of the possibility of such damage.

5.    CONTACT INFORMATION

Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany

www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------- */

/************************* System integration library **************************

   Author(s):   Manuel Jander

   Description:

*******************************************************************************/

/** \file   FDK_audio.h
 *  \brief  Global audio struct and constant definitions.
 */

#ifndef FDK_AUDIO_H
#define FDK_AUDIO_H

#include "machine_type.h"
#include "genericStds.h"
#include "syslib_channelMapDescr.h"

#ifdef __cplusplus
extern "C" {
#endif

/**
 * File format identifiers.
 */
typedef enum {
  FF_UNKNOWN = -1, /**< Unknown format.        */
  FF_RAW = 0,      /**< No container, bit stream data conveyed "as is". */

  FF_MP4_3GPP = 3, /**< 3GPP file format.      */
  FF_MP4_MP4F = 4, /**< MPEG-4 File format.     */

  FF_RAWPACKETS = 5 /**< Proprietary raw packet file. */

} FILE_FORMAT;

/**
 * Transport type identifiers.
 */
typedef enum {
  TT_UNKNOWN = -1, /**< Unknown format.            */
  TT_MP4_RAW = 0,  /**< "as is" access units (packet based since there is
                      obviously no sync layer) */
  TT_MP4_ADIF = 1, /**< ADIF bitstream format.     */
  TT_MP4_ADTS = 2, /**< ADTS bitstream format.     */

  TT_MP4_LATM_MCP1 = 6, /**< Audio Mux Elements with muxConfigPresent = 1 */
  TT_MP4_LATM_MCP0 = 7, /**< Audio Mux Elements with muxConfigPresent = 0, out
                           of band StreamMuxConfig */

  TT_MP4_LOAS = 10, /**< Audio Sync Stream.         */

  TT_DRM               = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
  TT_DABPLUS           = 13 /**< Digital Audio Broadcastong (DAB+) superframes bitstream format. */

} TRANSPORT_TYPE;

#define TT_IS_PACKET(x)                                                   \
  (((x) == TT_MP4_RAW) || ((x) == TT_DRM) || ((x) == TT_MP4_LATM_MCP0) || \
   ((x) == TT_MP4_LATM_MCP1))

/**
 * Audio Object Type definitions.
 */
typedef enum {
  AOT_NONE = -1,
  AOT_NULL_OBJECT = 0,
  AOT_AAC_MAIN = 1, /**< Main profile                              */
  AOT_AAC_LC = 2,   /**< Low Complexity object                     */
  AOT_AAC_SSR = 3,
  AOT_AAC_LTP = 4,
  AOT_SBR = 5,
  AOT_AAC_SCAL = 6,
  AOT_TWIN_VQ = 7,
  AOT_CELP = 8,
  AOT_HVXC = 9,
  AOT_RSVD_10 = 10,          /**< (reserved)                                */
  AOT_RSVD_11 = 11,          /**< (reserved)                                */
  AOT_TTSI = 12,             /**< TTSI Object                               */
  AOT_MAIN_SYNTH = 13,       /**< Main Synthetic object                     */
  AOT_WAV_TAB_SYNTH = 14,    /**< Wavetable Synthesis object                */
  AOT_GEN_MIDI = 15,         /**< General MIDI object                       */
  AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
  AOT_ER_AAC_LC = 17,        /**< Error Resilient(ER) AAC Low Complexity    */
  AOT_RSVD_18 = 18,          /**< (reserved)                                */
  AOT_ER_AAC_LTP = 19,       /**< Error Resilient(ER) AAC LTP object        */
  AOT_ER_AAC_SCAL = 20,      /**< Error Resilient(ER) AAC Scalable object   */
  AOT_ER_TWIN_VQ = 21,       /**< Error Resilient(ER) TwinVQ object         */
  AOT_ER_BSAC = 22,          /**< Error Resilient(ER) BSAC object           */
  AOT_ER_AAC_LD = 23,        /**< Error Resilient(ER) AAC LowDelay object   */
  AOT_ER_CELP = 24,          /**< Error Resilient(ER) CELP object           */
  AOT_ER_HVXC = 25,          /**< Error Resilient(ER) HVXC object           */
  AOT_ER_HILN = 26,          /**< Error Resilient(ER) HILN object           */
  AOT_ER_PARA = 27,          /**< Error Resilient(ER) Parametric object     */
  AOT_RSVD_28 = 28,          /**< might become SSC                          */
  AOT_PS = 29,               /**< PS, Parametric Stereo (includes SBR)      */
  AOT_MPEGS = 30,            /**< MPEG Surround                             */

  AOT_ESCAPE = 31, /**< Signal AOT uses more than 5 bits          */

  AOT_MP3ONMP4_L1 = 32, /**< MPEG-Layer1 in mp4                        */
  AOT_MP3ONMP4_L2 = 33, /**< MPEG-Layer2 in mp4                        */
  AOT_MP3ONMP4_L3 = 34, /**< MPEG-Layer3 in mp4                        */
  AOT_RSVD_35 = 35,     /**< might become DST                          */
  AOT_RSVD_36 = 36,     /**< might become ALS                          */
  AOT_AAC_SLS = 37,     /**< AAC + SLS                                 */
  AOT_SLS = 38,         /**< SLS                                       */
  AOT_ER_AAC_ELD = 39,  /**< AAC Enhanced Low Delay                    */

  AOT_USAC = 42,     /**< USAC                                      */
  AOT_SAOC = 43,     /**< SAOC                                      */
  AOT_LD_MPEGS = 44, /**< Low Delay MPEG Surround                   */

  AOT_DABPLUS_AAC_LC   = 135, /**< Virtual AOT for DAB plus AAC-LC                        */
  AOT_DABPLUS_SBR      = 136, /**< Virtual AOT for DAB plus HE-AAC                        */
  AOT_DABPLUS_PS       = 137, /**< Virtual AOT for DAB plus HE-AAC v2                     */


  /* Pseudo AOTs */
  AOT_MP2_AAC_LC = 129, /**< Virtual AOT MP2 Low Complexity profile */
  AOT_MP2_SBR = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR    */

  AOT_DRM_AAC = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR) */
  AOT_DRM_SBR = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR) */
  AOT_DRM_MPEG_PS =
      145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
  AOT_DRM_SURROUND =
      146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */
  AOT_DRM_USAC = 147 /**< Virtual AOT for DRM with USAC */

} AUDIO_OBJECT_TYPE;

#define CAN_DO_PS(aot)                                           \
  ((aot) == AOT_AAC_LC || (aot) == AOT_SBR || (aot) == AOT_PS || \
   (aot) == AOT_ER_BSAC || (aot) == AOT_DRM_AAC)

#define IS_USAC(aot) ((aot) == AOT_USAC)

#define IS_LOWDELAY(aot) ((aot) == AOT_ER_AAC_LD || (aot) == AOT_ER_AAC_ELD)

/** Channel Mode ( 1-7 equals MPEG channel configurations, others are
 * arbitrary). */
typedef enum {
  MODE_INVALID = -1,
  MODE_UNKNOWN = 0,
  MODE_1 = 1,         /**< C */
  MODE_2 = 2,         /**< L+R */
  MODE_1_2 = 3,       /**< C, L+R */
  MODE_1_2_1 = 4,     /**< C, L+R, Rear */
  MODE_1_2_2 = 5,     /**< C, L+R, LS+RS */
  MODE_1_2_2_1 = 6,   /**< C, L+R, LS+RS, LFE */
  MODE_1_2_2_2_1 = 7, /**< C, LC+RC, L+R, LS+RS, LFE */

  MODE_6_1 = 11,           /**< C, L+R, LS+RS, Crear, LFE */
  MODE_7_1_BACK = 12,      /**< C, L+R, LS+RS, Lrear+Rrear, LFE */
  MODE_7_1_TOP_FRONT = 14, /**< C, L+R, LS+RS, LFE, Ltop+Rtop */

  MODE_7_1_REAR_SURROUND = 33, /**< C, L+R, LS+RS, Lrear+Rrear, LFE */
  MODE_7_1_FRONT_CENTER = 34,  /**< C, LC+RC, L+R, LS+RS, LFE */

  MODE_212 = 128 /**< 212 configuration, used in ELDv2 */

} CHANNEL_MODE;

/**
 * Speaker description tags.
 * Do not change the enumeration values unless it keeps the following
 * segmentation:
 * - Bit 0-3: Horizontal postion (0: none, 1: front, 2: side, 3: back, 4: lfe)
 * - Bit 4-7: Vertical position (0: normal, 1: top, 2: bottom)
 */
typedef enum {
  ACT_NONE = 0x00,
  ACT_FRONT = 0x01, /*!< Front speaker position (at normal height) */
  ACT_SIDE = 0x02,  /*!< Side speaker position (at normal height) */
  ACT_BACK = 0x03,  /*!< Back speaker position (at normal height) */
  ACT_LFE = 0x04,   /*!< Low frequency effect speaker postion (front) */

  ACT_TOP =
      0x10, /*!< Top speaker area (for combination with speaker positions) */
  ACT_FRONT_TOP = 0x11, /*!< Top front speaker = (ACT_FRONT|ACT_TOP) */
  ACT_SIDE_TOP = 0x12,  /*!< Top side speaker  = (ACT_SIDE |ACT_TOP) */
  ACT_BACK_TOP = 0x13,  /*!< Top back speaker  = (ACT_BACK |ACT_TOP) */

  ACT_BOTTOM =
      0x20, /*!< Bottom speaker area (for combination with speaker positions) */
  ACT_FRONT_BOTTOM = 0x21, /*!< Bottom front speaker = (ACT_FRONT|ACT_BOTTOM) */
  ACT_SIDE_BOTTOM = 0x22,  /*!< Bottom side speaker  = (ACT_SIDE |ACT_BOTTOM) */
  ACT_BACK_BOTTOM = 0x23   /*!< Bottom back speaker  = (ACT_BACK |ACT_BOTTOM) */

} AUDIO_CHANNEL_TYPE;

typedef enum {
  SIG_UNKNOWN = -1,
  SIG_IMPLICIT = 0,
  SIG_EXPLICIT_BW_COMPATIBLE = 1,
  SIG_EXPLICIT_HIERARCHICAL = 2

} SBR_PS_SIGNALING;

/**
 * Audio Codec flags.
 */
#define AC_ER_VCB11                                                           \
  0x000001 /*!< aacSectionDataResilienceFlag     flag (from ASC): 1 means use \
              virtual codebooks  */
#define AC_ER_RVLC                                                             \
  0x000002 /*!< aacSpectralDataResilienceFlag     flag (from ASC): 1 means use \
              huffman codeword reordering */
#define AC_ER_HCR                                                             \
  0x000004 /*!< aacSectionDataResilienceFlag     flag (from ASC): 1 means use \
              virtual codebooks  */
#define AC_SCALABLE 0x000008    /*!< AAC Scalable*/
#define AC_ELD 0x000010         /*!< AAC-ELD */
#define AC_LD 0x000020          /*!< AAC-LD */
#define AC_ER 0x000040          /*!< ER syntax */
#define AC_BSAC 0x000080        /*!< BSAC */
#define AC_USAC 0x000100        /*!< USAC */
#define AC_RSV603DA 0x000200    /*!< RSVD60 3D audio */
#define AC_HDAAC 0x000400       /*!< HD-AAC */
#define AC_RSVD50 0x004000      /*!< Rsvd50 */
#define AC_SBR_PRESENT 0x008000 /*!< SBR present flag (from ASC) */
#define AC_SBRCRC \
  0x010000 /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */
#define AC_PS_PRESENT 0x020000 /*!< PS present flag (from ASC or implicit)  */
#define AC_MPS_PRESENT                                                     \
  0x040000                    /*!< MPS present flag (from ASC or implicit) \
                               */
#define AC_DRM 0x080000       /*!< DRM bit stream syntax */
#define AC_INDEP 0x100000     /*!< Independency flag */
#define AC_MPEGD_RES 0x200000 /*!< MPEG-D residual individual channel data. */
#define AC_SAOC_PRESENT 0x400000   /*!< SAOC Present Flag */
#define AC_DAB 0x800000            /*!< DAB bit stream syntax */
#define AC_ELD_DOWNSCALE 0x1000000 /*!< ELD Downscaled playout */
#define AC_LD_MPS 0x2000000        /*!< Low Delay MPS. */
#define AC_DRC_PRESENT                                   \
  0x4000000 /*!< Dynamic Range Control (DRC) data found. \
             */
#define AC_USAC_SCFGI3 \
  0x8000000 /*!< USAC flag: If stereoConfigIndex is 3 the flag is set. */
/**
 * Audio Codec flags (reconfiguration).
 */
#define AC_CM_DET_CFG_CHANGE                                                 \
  0x000001 /*!< Config mode signalizes the callback to work in config change \
              detection mode */
#define AC_CM_ALLOC_MEM                                               \
  0x000002 /*!< Config mode signalizes the callback to work in memory \
              allocation mode */

/**
 * Audio Codec flags (element specific).
 */
#define AC_EL_USAC_TW 0x000001    /*!< USAC time warped filter bank is active */
#define AC_EL_USAC_NOISE 0x000002 /*!< USAC noise filling is active */
#define AC_EL_USAC_ITES 0x000004  /*!< USAC SBR inter-TES tool is active */
#define AC_EL_USAC_PVC \
  0x000008 /*!< USAC SBR predictive vector coding tool is active */
#define AC_EL_USAC_MPS212 0x000010 /*!< USAC MPS212 tool is active */
#define AC_EL_USAC_LFE 0x000020    /*!< USAC element is LFE */
#define AC_EL_USAC_CP_POSSIBLE                                                 \
  0x000040 /*!< USAC may use Complex Stereo Prediction in this channel element \
            */
#define AC_EL_ENHANCED_NOISE 0x000080   /*!< Enhanced noise filling*/
#define AC_EL_IGF_AFTER_TNS 0x000100    /*!< IGF after TNS */
#define AC_EL_IGF_INDEP_TILING 0x000200 /*!< IGF independent tiling */
#define AC_EL_IGF_USE_ENF 0x000400      /*!< IGF use enhanced noise filling */
#define AC_EL_FULLBANDLPD 0x000800      /*!< enable fullband LPD tools */
#define AC_EL_LPDSTEREOIDX 0x001000     /*!< LPD-stereo-tool stereo index */
#define AC_EL_LFE 0x002000              /*!< The element is of type LFE. */

/* CODER_CONFIG::flags */
#define CC_MPEG_ID 0x00100000
#define CC_IS_BASELAYER 0x00200000
#define CC_PROTECTION 0x00400000
#define CC_SBR 0x00800000
#define CC_SBRCRC 0x00010000
#define CC_SAC 0x00020000
#define CC_RVLC 0x01000000
#define CC_VCB11 0x02000000
#define CC_HCR 0x04000000
#define CC_PSEUDO_SURROUND 0x08000000
#define CC_USAC_NOISE 0x10000000
#define CC_USAC_TW 0x20000000
#define CC_USAC_HBE 0x40000000

/** Generic audio coder configuration structure. */
typedef struct {
  AUDIO_OBJECT_TYPE aot;     /**< Audio Object Type (AOT).           */
  AUDIO_OBJECT_TYPE extAOT;  /**< Extension Audio Object Type (SBR). */
  CHANNEL_MODE channelMode;  /**< Channel mode.                      */
  UCHAR channelConfigZero;   /**< Use channel config zero + pce although a
                                standard channel config could be signaled. */
  INT samplingRate;          /**< Sampling rate.                     */
  INT extSamplingRate;       /**< Extended samplerate (SBR).         */
  INT downscaleSamplingRate; /**< Downscale sampling rate (ELD downscaled mode)
                              */
  INT bitRate;               /**< Average bitrate.                   */
  int samplesPerFrame; /**< Number of PCM samples per codec frame and audio
                          channel. */
  int noChannels;      /**< Number of audio channels.          */
  int bitsFrame;
  int nSubFrames; /**< Amount of encoder subframes. 1 means no subframing. */
  int BSACnumOfSubFrame; /**< The number of the sub-frames which are grouped and
                            transmitted in a super-frame (BSAC). */
  int BSAClayerLength; /**< The average length of the large-step layers in bytes
                          (BSAC).                            */
  UINT flags;          /**< flags */
  UCHAR matrixMixdownA; /**< Matrix mixdown index to put into PCE. Default value
                           0 means no mixdown coefficient, valid values are 1-4
                           which correspond to matrix_mixdown_idx 0-3. */
  UCHAR headerPeriod;   /**< Frame period for sending in band configuration
                           buffers in the transport layer. */

  UCHAR stereoConfigIndex;       /**< USAC MPS stereo mode */
  UCHAR sbrMode;                 /**< USAC SBR mode */
  SBR_PS_SIGNALING sbrSignaling; /**< 0: implicit signaling, 1: backwards
                                    compatible explicit signaling, 2:
                                    hierarcical explicit signaling */

  UCHAR rawConfig[64]; /**< raw codec specific config as bit stream */
  int rawConfigBits;   /**< Size of rawConfig in bits */

  UCHAR sbrPresent;
  UCHAR psPresent;
} CODER_CONFIG;

#define USAC_ID_BIT 16 /** USAC element IDs start at USAC_ID_BIT */

/** MP4 Element IDs. */
typedef enum {
  /* mp4 element IDs */
  ID_NONE = -1, /**< Invalid Element helper ID.             */
  ID_SCE = 0,   /**< Single Channel Element.                */
  ID_CPE = 1,   /**< Channel Pair Element.                  */
  ID_CCE = 2,   /**< Coupling Channel Element.              */
  ID_LFE = 3,   /**< LFE Channel Element.                   */
  ID_DSE = 4,   /**< Currently one Data Stream Element for ancillary data is
                   supported. */
  ID_PCE = 5,   /**< Program Config Element.                */
  ID_FIL = 6,   /**< Fill Element.                          */
  ID_END = 7,   /**< Arnie (End Element = Terminator).      */
  ID_EXT = 8,   /**< Extension Payload (ER only).           */
  ID_SCAL = 9,  /**< AAC scalable element (ER only).        */
  /* USAC element IDs */
  ID_USAC_SCE = 0 + USAC_ID_BIT, /**< Single Channel Element.                */
  ID_USAC_CPE = 1 + USAC_ID_BIT, /**< Channel Pair Element.                  */
  ID_USAC_LFE = 2 + USAC_ID_BIT, /**< LFE Channel Element.                   */
  ID_USAC_EXT = 3 + USAC_ID_BIT, /**< Extension Element.                     */
  ID_USAC_END = 4 + USAC_ID_BIT, /**< Arnie (End Element = Terminator).      */
  ID_LAST
} MP4_ELEMENT_ID;

/* usacConfigExtType q.v. ISO/IEC DIS 23008-3 Table 52  and  ISO/IEC FDIS
 * 23003-3:2011(E) Table 74*/
typedef enum {
  /* USAC and RSVD60 3DA */
  ID_CONFIG_EXT_FILL = 0,
  /* RSVD60 3DA */
  ID_CONFIG_EXT_DOWNMIX = 1,
  ID_CONFIG_EXT_LOUDNESS_INFO = 2,
  ID_CONFIG_EXT_AUDIOSCENE_INFO = 3,
  ID_CONFIG_EXT_HOA_MATRIX = 4,
  ID_CONFIG_EXT_SIG_GROUP_INFO = 6
  /* 5-127 => reserved for ISO use */
  /* > 128 => reserved for use outside of ISO scope */
} CONFIG_EXT_ID;

#define IS_CHANNEL_ELEMENT(elementId)                                         \
  ((elementId) == ID_SCE || (elementId) == ID_CPE || (elementId) == ID_LFE || \
   (elementId) == ID_USAC_SCE || (elementId) == ID_USAC_CPE ||                \
   (elementId) == ID_USAC_LFE)

#define IS_MP4_CHANNEL_ELEMENT(elementId) \
  ((elementId) == ID_SCE || (elementId) == ID_CPE || (elementId) == ID_LFE)

#define EXT_ID_BITS 4 /**< Size in bits of extension payload type tags. */

/** Extension payload types. */
typedef enum {
  EXT_FIL = 0x00,
  EXT_FILL_DATA = 0x01,
  EXT_DATA_ELEMENT = 0x02,
  EXT_DATA_LENGTH = 0x03,
  EXT_UNI_DRC = 0x04,
  EXT_LDSAC_DATA = 0x09,
  EXT_SAOC_DATA = 0x0a,
  EXT_DYNAMIC_RANGE = 0x0b,
  EXT_SAC_DATA = 0x0c,
  EXT_SBR_DATA = 0x0d,
  EXT_SBR_DATA_CRC = 0x0e
} EXT_PAYLOAD_TYPE;

#define IS_USAC_CHANNEL_ELEMENT(elementId)                     \
  ((elementId) == ID_USAC_SCE || (elementId) == ID_USAC_CPE || \
   (elementId) == ID_USAC_LFE)

/** MPEG-D USAC & RSVD60 3D audio Extension Element Types. */
typedef enum {
  /* usac */
  ID_EXT_ELE_FILL = 0x00,
  ID_EXT_ELE_MPEGS = 0x01,
  ID_EXT_ELE_SAOC = 0x02,
  ID_EXT_ELE_AUDIOPREROLL = 0x03,
  ID_EXT_ELE_UNI_DRC = 0x04,
  /* rsv603da */
  ID_EXT_ELE_OBJ_METADATA = 0x05,
  ID_EXT_ELE_SAOC_3D = 0x06,
  ID_EXT_ELE_HOA = 0x07,
  ID_EXT_ELE_FMT_CNVRTR = 0x08,
  ID_EXT_ELE_MCT = 0x09,
  ID_EXT_ELE_ENHANCED_OBJ_METADATA = 0x0d,
  /* reserved for use outside of ISO scope */
  ID_EXT_ELE_VR_METADATA = 0x81,
  ID_EXT_ELE_UNKNOWN = 0xFF
} USAC_EXT_ELEMENT_TYPE;

/**
 * Proprietary raw packet file configuration data type identifier.
 */
typedef enum {
  TC_NOTHING = 0,  /* No configuration available -> in-band configuration.   */
  TC_RAW_ADTS = 2, /* Transfer type is ADTS. */
  TC_RAW_LATM_MCP1 = 6, /* Transfer type is LATM with SMC present.    */
  TC_RAW_SDC = 21       /* Configuration data field is Drm SDC.             */

} TP_CONFIG_TYPE;

/*
 * ##############################################################################################
 * Library identification and error handling
 * ##############################################################################################
 */
/* \cond */

typedef enum {
  FDK_NONE = 0,
  FDK_TOOLS = 1,
  FDK_SYSLIB = 2,
  FDK_AACDEC = 3,
  FDK_AACENC = 4,
  FDK_SBRDEC = 5,
  FDK_SBRENC = 6,
  FDK_TPDEC = 7,
  FDK_TPENC = 8,
  FDK_MPSDEC = 9,
  FDK_MPEGFILEREAD = 10,
  FDK_MPEGFILEWRITE = 11,
  FDK_PCMDMX = 31,
  FDK_MPSENC = 34,
  FDK_TDLIMIT = 35,
  FDK_UNIDRCDEC = 38,

  FDK_MODULE_LAST

} FDK_MODULE_ID;

/* AAC capability flags */
#define CAPF_AAC_LC 0x00000001 /**< Support flag for AAC Low Complexity. */
#define CAPF_ER_AAC_LD                                                        \
  0x00000002 /**< Support flag for AAC Low Delay with Error Resilience tools. \
              */
#define CAPF_ER_AAC_SCAL 0x00000004 /**< Support flag for AAC Scalable. */
#define CAPF_ER_AAC_LC                                                      \
  0x00000008 /**< Support flag for AAC Low Complexity with Error Resilience \
                tools. */
#define CAPF_AAC_480 \
  0x00000010 /**< Support flag for AAC with 480 framelength.  */
#define CAPF_AAC_512 \
  0x00000020 /**< Support flag for AAC with 512 framelength.  */
#define CAPF_AAC_960 \
  0x00000040 /**< Support flag for AAC with 960 framelength.  */
#define CAPF_AAC_1024 \
  0x00000080 /**< Support flag for AAC with 1024 framelength. */
#define CAPF_AAC_HCR \
  0x00000100 /**< Support flag for AAC with Huffman Codeword Reordering.    */
#define CAPF_AAC_VCB11 \
  0x00000200 /**< Support flag for AAC Virtual Codebook 11.    */
#define CAPF_AAC_RVLC \
  0x00000400 /**< Support flag for AAC Reversible Variable Length Coding.   */
#define CAPF_AAC_MPEG4 0x00000800 /**< Support flag for MPEG file format. */
#define CAPF_AAC_DRC \
  0x00001000 /**< Support flag for AAC Dynamic Range Control. */
#define CAPF_AAC_CONCEALMENT \
  0x00002000 /**< Support flag for AAC concealment.           */
#define CAPF_AAC_DRM_BSFORMAT \
  0x00004000 /**< Support flag for AAC DRM bistream format. */
#define CAPF_ER_AAC_ELD                                              \
  0x00008000 /**< Support flag for AAC Enhanced Low Delay with Error \
                Resilience tools.  */
#define CAPF_ER_AAC_BSAC \
  0x00010000 /**< Support flag for AAC BSAC.                           */
#define CAPF_AAC_ELD_DOWNSCALE \
  0x00040000 /**< Support flag for AAC-ELD Downscaling           */
#define CAPF_AAC_USAC_LP \
  0x00100000 /**< Support flag for USAC low power mode. */
#define CAPF_AAC_USAC \
  0x00200000 /**< Support flag for Unified Speech and Audio Coding (USAC). */
#define CAPF_ER_AAC_ELDV2 \
  0x00800000 /**< Support flag for AAC Enhanced Low Delay with MPS 212.  */
#define CAPF_AAC_UNIDRC \
  0x01000000 /**< Support flag for MPEG-D Dynamic Range Control (uniDrc). */

/* Transport capability flags */
#define CAPF_ADTS \
  0x00000001 /**< Support flag for ADTS transport format.        */
#define CAPF_ADIF \
  0x00000002 /**< Support flag for ADIF transport format.        */
#define CAPF_LATM \
  0x00000004 /**< Support flag for LATM transport format.        */
#define CAPF_LOAS \
  0x00000008 /**< Support flag for LOAS transport format.        */
#define CAPF_RAWPACKETS \
  0x00000010 /**< Support flag for RAW PACKETS transport format. */
#define CAPF_DRM \
  0x00000020 /**< Support flag for DRM/DRM+ transport format.    */
#define CAPF_RSVD50 \
  0x00000040 /**< Support flag for RSVD50 transport format       */

/* SBR capability flags */
#define CAPF_SBR_LP \
  0x00000001 /**< Support flag for SBR Low Power mode.           */
#define CAPF_SBR_HQ \
  0x00000002 /**< Support flag for SBR High Quality mode.        */
#define CAPF_SBR_DRM_BS \
  0x00000004 /**< Support flag for                               */
#define CAPF_SBR_CONCEALMENT \
  0x00000008 /**< Support flag for SBR concealment.              */
#define CAPF_SBR_DRC \
  0x00000010 /**< Support flag for SBR Dynamic Range Control.    */
#define CAPF_SBR_PS_MPEG \
  0x00000020 /**< Support flag for MPEG Parametric Stereo.       */
#define CAPF_SBR_PS_DRM \
  0x00000040 /**< Support flag for DRM Parametric Stereo.        */
#define CAPF_SBR_ELD_DOWNSCALE \
  0x00000080 /**< Support flag for ELD reduced delay mode        */
#define CAPF_SBR_HBEHQ \
  0x00000100 /**< Support flag for HQ HBE                        */

/* DAB capability flags */
#define CAPF_DAB_MP2          0x00000001  /**< Support flag for Layer2 DAB.                   */
#define CAPF_DAB_AAC          0x00000002  /**< Support flag for DAB+ (HE-AAC v2).             */
#define CAPF_DAB_PAD          0x00000004  /**< Support flag for PAD extraction.               */
#define CAPF_DAB_DRC          0x00000008  /**< Support flag for Dynamic Range Control.        */
#define CAPF_DAB_SURROUND     0x00000010  /**< Support flag for DAB Surround (MPS).           */


/* PCM utils capability flags */
#define CAPF_DMX_BLIND \
  0x00000001 /**< Support flag for blind downmixing.             */
#define CAPF_DMX_PCE                                                      \
  0x00000002 /**< Support flag for guided downmix with data from MPEG-2/4 \
                Program Config Elements (PCE). */
#define CAPF_DMX_ARIB                                                         \
  0x00000004 /**< Support flag for PCE guided downmix with slightly different \
                equations and levels to fulfill ARIB standard. */
#define CAPF_DMX_DVB                                                           \
  0x00000008 /**< Support flag for guided downmix with data from DVB ancillary \
                data fields. */
#define CAPF_DMX_CH_EXP                                                       \
  0x00000010 /**< Support flag for simple upmixing by dublicating channels or \
                adding zero channels. */
#define CAPF_DMX_6_CH                                                   \
  0x00000020 /**< Support flag for 5.1 channel configuration (input and \
                output). */
#define CAPF_DMX_8_CH                                                          \
  0x00000040 /**< Support flag for 6 and 7.1 channel configurations (input and \
                output). */
#define CAPF_DMX_24_CH                                                   \
  0x00000080 /**< Support flag for 22.2 channel configuration (input and \
                output). */
#define CAPF_LIMITER                                      \
  0x00002000 /**< Support flag for signal level limiting. \
              */

/* MPEG Surround capability flags */
#define CAPF_MPS_STD \
  0x00000001 /**< Support flag for MPEG Surround.           */
#define CAPF_MPS_LD                                         \
  0x00000002 /**< Support flag for Low Delay MPEG Surround. \
              */
#define CAPF_MPS_USAC \
  0x00000004 /**< Support flag for USAC MPEG Surround.      */
#define CAPF_MPS_HQ                                                     \
  0x00000010 /**< Support flag indicating if high quality processing is \
                supported */
#define CAPF_MPS_LP                                                        \
  0x00000020 /**< Support flag indicating if partially complex (low power) \
                processing is supported */
#define CAPF_MPS_BLIND \
  0x00000040 /**< Support flag indicating if blind processing is supported */
#define CAPF_MPS_BINAURAL \
  0x00000080 /**< Support flag indicating if binaural output is possible */
#define CAPF_MPS_2CH_OUT \
  0x00000100 /**< Support flag indicating if 2ch output is possible      */
#define CAPF_MPS_6CH_OUT \
  0x00000200 /**< Support flag indicating if 6ch output is possible      */
#define CAPF_MPS_8CH_OUT \
  0x00000400 /**< Support flag indicating if 8ch output is possible      */
#define CAPF_MPS_1CH_IN \
  0x00001000 /**< Support flag indicating if 1ch dmx input is possible   */
#define CAPF_MPS_2CH_IN \
  0x00002000 /**< Support flag indicating if 2ch dmx input is possible   */
#define CAPF_MPS_6CH_IN \
  0x00004000 /**< Support flag indicating if 5ch dmx input is possible   */

/* \endcond */

/*
 * ##############################################################################################
 * Library versioning
 * ##############################################################################################
 */

/**
 * Convert each member of version numbers to one single numeric version
 * representation.
 * \param lev0  1st level of version number.
 * \param lev1  2nd level of version number.
 * \param lev2  3rd level of version number.
 */
#define LIB_VERSION(lev0, lev1, lev2)                      \
  ((lev0 << 24 & 0xff000000) | (lev1 << 16 & 0x00ff0000) | \
   (lev2 << 8 & 0x0000ff00))

/**
 *  Build text string of version.
 */
#define LIB_VERSION_STRING(info)                                               \
  FDKsprintf((info)->versionStr, "%d.%d.%d", (((info)->version >> 24) & 0xff), \
             (((info)->version >> 16) & 0xff),                                 \
             (((info)->version >> 8) & 0xff))

/**
 *  Library information.
 */
typedef struct LIB_INFO {
  const char* title;
  const char* build_date;
  const char* build_time;
  FDK_MODULE_ID module_id;
  INT version;
  UINT flags;
  char versionStr[32];
} LIB_INFO;

#ifdef __cplusplus
#define FDK_AUDIO_INLINE inline
#else
#define FDK_AUDIO_INLINE
#endif

/** Initialize library info. */
static FDK_AUDIO_INLINE void FDKinitLibInfo(LIB_INFO* info) {
  int i;

  for (i = 0; i < FDK_MODULE_LAST; i++) {
    info[i].module_id = FDK_NONE;
  }
}

/** Aquire supported features of library. */
static FDK_AUDIO_INLINE UINT
FDKlibInfo_getCapabilities(const LIB_INFO* info, FDK_MODULE_ID module_id) {
  int i;

  for (i = 0; i < FDK_MODULE_LAST; i++) {
    if (info[i].module_id == module_id) {
      return info[i].flags;
    }
  }
  return 0;
}

/** Search for next free tab. */
static FDK_AUDIO_INLINE INT FDKlibInfo_lookup(const LIB_INFO* info,
                                              FDK_MODULE_ID module_id) {
  int i = -1;

  for (i = 0; i < FDK_MODULE_LAST; i++) {
    if (info[i].module_id == module_id) return -1;
    if (info[i].module_id == FDK_NONE) break;
  }
  if (i == FDK_MODULE_LAST) return -1;

  return i;
}

/*
 * ##############################################################################################
 * Buffer description
 * ##############################################################################################
 */

/**
 *  I/O buffer descriptor.
 */
typedef struct FDK_bufDescr {
  void** ppBase;  /*!< Pointer to an array containing buffer base addresses.
                       Set to NULL for buffer requirement info. */
  UINT* pBufSize; /*!< Pointer to an array containing the number of elements
                     that can be placed in the specific buffer. */
  UINT* pEleSize; /*!< Pointer to an array containing the element size for each
                     buffer in bytes. That is mostly the number returned by the
                     sizeof() operator for the data type used for the specific
                     buffer. */
  UINT*
      pBufType; /*!< Pointer to an array of bit fields containing a description
                     for each buffer. See XXX below for more details.  */
  UINT numBufs; /*!< Total number of buffers. */

} FDK_bufDescr;

/**
 * Buffer type description field.
 */
#define FDK_BUF_TYPE_MASK_IO ((UINT)0x03 << 30)
#define FDK_BUF_TYPE_MASK_DESCR ((UINT)0x3F << 16)
#define FDK_BUF_TYPE_MASK_ID ((UINT)0xFF)

#define FDK_BUF_TYPE_INPUT ((UINT)0x1 << 30)
#define FDK_BUF_TYPE_OUTPUT ((UINT)0x2 << 30)

#define FDK_BUF_TYPE_PCM_DATA ((UINT)0x1 << 16)
#define FDK_BUF_TYPE_ANC_DATA ((UINT)0x2 << 16)
#define FDK_BUF_TYPE_BS_DATA ((UINT)0x4 << 16)

#ifdef __cplusplus
}
#endif

#endif /* FDK_AUDIO_H */